Files
asterisk/configs
Brett Bryant 12d5cebea2 Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on 
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.

(issue #12799)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-27 16:28:06 +00:00
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